593e32551c
Previously, the RTCStatsCollector needed to ask the voice/video channel for its transport name in order to generate transport level stats. That would happen on the networking thread which was unsafe because the voice/video channel could have disappeared in the duration of the asynchronous thread hop from the signaling thread to the networking thread. This changes the networking stats code to check a saved map that tracks the transport name for each voice/video channel. Bug: None Change-Id: I1f03ba8c0526eaa4419f660f18b8b9da62c3f932 Reviewed-on: https://webrtc-review.googlesource.com/33660 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21332}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct